Unit-4

Transport Layer
Transport Services
Elements of Transport protocols
Connection management
TCP and UDP protocols.
 Introduction of Transport layer
 Transport Layer is the second layer in the TCP/IP model and the
fourth layer in the OSI model.
 The unit of data encapsulation in the Transport Layer is a segment.
 The Transport Layer is not just another layer, it is the heart of the
whole protocol hierarchy.
 Its task is to provide reliable, cost-effective data transport from the
source machine to destination machine, independently of the
physical network or network’s currently in use.
 Without the transport layer, the whole concept of layered protocols
would make little sense.
 Communication is provided using a logical connection, which means
that the two application layers, which can be located in different
parts of the globe, assume that there is an imaginary direct
connection through which they can send and receive messages.
 Transport layer Services
 The main role of the transport layer is to provide
the communication services directly to the
application processes running on different hosts.
 The transport layer protocols are implemented in
the end systems but not in the network routers.
 A computer network provides more than one protocol
to the network applications. For example, TCP and
UDP are two transport layer protocols that provide a
different set of services to the network layer.
 All transport layer protocols provide
multiplexing/demultiplexing service.
 It also provides other services such as reliable data
transfer, bandwidth guarantees, and delay guarantees.
 Each of the applications in the application layer has
the ability to send a message by using TCP or UDP.
 The application communicates by using either of
these two protocols. Both TCP and UDP will then
communicate with the internet protocol in the
internet layer.
 The applications can read and write to the transport
layer. Therefore, we can say that communication is a
two-way process.
Services Provided to the Upper Layer:
 The ultimate goal of the transport layer is to provide efficient,
reliable, and cost-effective services to its users, normally
processes in the application layer.
 To achieve this goal, the Transport layer makes use of the
services provided by the network layer.
 The hardware or software within the transport layer that
does the work is called the Transport entity.
 The transport entity can be located in the OS kernel .
 The relationship of the network, transport & application
layers is shown in below figure
 At the sender’s side: The transport layer receives
data (message) from the Application layer and then
performs Segmentation, divides the actual message
into segments, adds source and destination’s port
numbers into the header of the segment, and
transfers the message to the Network layer.
 At the receiver’s side: The transport layer receives
data from the Network layer, reassembles the
segmented data, reads its header, identifies the port
number, and forwards the message to the
appropriate port in the Application layer.
 In network layer, there are two types of network service,
connection-oriented and connectionless, there are also two
types of transport service. The connection-oriented
transport service is similar to the connection-oriented
network service in many ways. The connectionless
transport service is also very similar to the connectionless
network service.
 In both cases, connections have three phases:
establishment, data transfer, and release.
 Addressing and flow control are also similar in both layers.
 The transport code runs entirely on the users machines ,
but the network layer mostly runs on the routers, which
are operated by the carrier.
Transport Service Primitives
 To allow users to access the transport service, the transport
layer must provide some operations to application programs,
that is, a transport service interface. Each transport service
has its own interface.
 The transport service is similar to the network service, but
there are also some important differences. The main
difference is that the network service is intended to model
the service offered by real networks and all. Real networks
can lose packets, so the network service is generally
unreliable.
 The connection-oriented transport service, in contrast, is
reliable. Of course, real networks are not error-free, but that
is precisely the purpose of the transport layer—to provide a
reliable service on top of an unreliable network.
 A second difference between the network service and
transport service is that the services are intended for. The
network service is used only by the transport entities. Few
users write their own transport entities, and thus few users or
programs ever (meaning always/forever/still) see the bare
network service.
 To get an idea of what a transport service might be like,
consider the five primitives listed in below Fig.
Figure : The primitives for a simple transport service.

Primitive Packet sent Meaning

LISTEN (none) Block until some process tries

to connect
CONNEC CONNECTION Actively attempt to establish a
T REQ. connection
SEND DATA Send information

RECEIVE (none) Block until a DATA packet

arrives
DISCONN DISCONNECTIO Request a release of the
ECT N REQ. connection
 TPDU (Transport Protocol Data Unit) for messages sent
from transport entity to transport entity.
 TPDU are contained in packets. Packets are contained in
frames (exchanged by the data link layer).
 When a frame arrives, the data link layer processes the
frame header and, if the destination address matches for
local delivery, passes the contents of the frame payload field
up to the network entity.
 The network entity similarly processes the packet header and
then passes the contents of the packet payload up to the
transport entity.
 This nesting is illustrated in below Fig.
 A state diagram for connection establishment and release for
these simple primitives is given in below Fig.
 Each transition is triggered by some event, either a
primitive executed by the local transport user or an incoming
packet.
 For simplicity, we assume here that each segment is
separately acknowledged. We also assume that a symmetric
disconnection model is used, with the client going first.
Please note that this model is quite unsophisticated.
Figure: A state diagram for a simple connection management scheme.
Transitions labeled in italics are caused by packet arrivals. The solid lines
show the client’s state sequence. The dashed lines show the server’s
state sequence
 Elements of Transport protocols
The transport service is implemented by a transport
protocol used between the two transport entities.
In some ways, transport protocols resemble the data link
protocols. Both have to deal with error control, sequencing,
and flow control, among other issues.
However, significant differences between the two also exist.
These differences are due to major dissimilarities between
the environments in which the two protocols operate, as
shown in below Fig.
Figure: Environment of the (a) data link layer (b) transport layer
At the data link layer, two routers communicate directly via
a physical channel, whether wired or wireless,
whereas at the transport layer, this physical channel is
replaced by the entire network.
For one thing, over point-to-point links such as wires or
optical fiber, it is usually not necessary for a router to
specify which router it wants to talk to—each outgoing
line leads directly to a particular router.
In the transport layer, explicit addressing of destinations
is required.
 For another thing, the process of establishing a connection
over the wire of Fig.(a) is simple.The other end is always
there (unless it has crashed, in which case it is not there).
Either way, there is not much to do.
 Even on wireless links, the process is not much different.
Just sending a message is sufficient to have it reach all other
destinations. If the message is not acknowledged due to an
error, it can be resent. In the transport layer, initial
connection establishment is complicated.
1. Addressing:
 When an application (e.g., a user) process wishes to set up
a connection to a remote application process, it must
specify which one to connect to. (Connectionless transport
has the same problem: to whom should each message be
sent?) The method normally used is to define transport
addresses to which processes can listen for connection
requests.
 In the Internet, these endpoints are called ports. We will
use the generic term TSAP (Transport Service Access
Point) to mean a specific endpoint in the transport layer.
In Internet, they consists of an (IP address, local port) pair.
 The analogous endpoints in the network layer (i.e.,
network layer addresses) are naturally called NSAPs
(Network Service Access Points). IP addresses are
Figure: illustrates the relationship between the
NSAPs, the TSAPs, and a transport connection.
 Application processes, both clients and servers, can attach
themselves to a local TSAP to establish a connection to a
remote TSAP. These connections run through NSAPs on
each host, as shown in figure.
 A possible scenario for a transport connection is as follows:
1. A mail server process attaches itself to TSAP 1522 on host 2 to
wait for an incoming call. A call such as our LISTEN might be
used, for example.
2. An application process on host 1 wants to send an email message,
so it attaches itself to TSAP 1208 and issues a CONNECT request.
The request specifies TSAP 1208 on host 1 as the source and
TSAP 1522 on host 2 as the destination. This action ultimately
results in a transport connection being established between the
application process and the server.
3. The application process sends over the mail message.
4. The mail server responds to say that it will deliver
the message.
5. The transport connection is released.
2. Connection Establishment:
 Establishing a connection sounds easy, but it is actually
surprisingly tricky. At first , it would seem sufficient for one
transport entity to just send a CONNECTION REQUEST
segment to the destination and wait for a CONNECTION
ACCEPTED reply. The problem occurs when the network
can lose, delay, corrupt, and duplicate packets. This
behavior causes serious complications.
 Imagine a network that is so congested that
acknowledgements hardly ever get back in time and each
packet times out and is retransmitted two or three times.
Suppose that the network uses datagram inside and that
every packet follows a different route.
 Some of the packets might get stuck in a traffic jam
inside the network and take a long time to arrive. That is,
they may be delayed in the network and pop out much later,
when the sender thought that they had been lost.
 The worst possible nightmare is as follows. “A user
establishes a connection with a bank, sends messages
telling the bank to transfer a large amount of money to
the account of a not-entirelytrustworthy person “.
Unfortunately, the packets decide to take the scenic route to
the destination and go off exploring a remote corner of the
network.
 The sender then times out and sends them all again. This
time the packets take the shortest route and are delivered
quickly so the sender releases the connection.
 Unfortunately, eventually the initial batch of packets finally
come out of hiding and arrives at the destination in order,
asking the bank to establish a new connection and transfer
money (again). The bank has no way of telling that these are
duplicates. It must assume that this is a second, independent
transaction, and transfers the money again.
 The root of the problem is that the delayed duplicates
are thought to be new packets. We cannot prevent
packets from being duplicated and delayed. But if and
when this happens, the packets must be rejected as
duplicates and not processed as fresh packets.
 The problem can be attacked in various ways, none of them
very satisfactory. One way is to use throwaway transport
addresses. In this approach, each time a transport address is
needed, a new one is generated. When a connection is
released, the address is discarded and never used again.
Delayed duplicate packets then never find their way to a
transport process and can do no damage.
Note: However, this approach makes it more difficult to
connect with a process in the first place.
 Another possibility is to give each connection a unique
identifier (i.e., a sequence number incremented for each
connection established) chosen by the initiating party and
put in each segment, including the one requesting the
connection
 After each connection is released, each transport entity can
update a table listing obsolete connections as (peer transport
entity, connection identifier) pairs. Whenever a connection
request comes in, it can be checked against the table to see if
it belongs to a previously released connection.
 Unfortunately, this scheme has a basic flaw: it requires each
transport entity to maintain a certain amount of history
information indefinitely. This history must persist at both the
source and destination machines. Otherwise, if a machine
crashes and loses its memory, it will no longer know which
connection identifiers have already been used by its peers.
 Instead, we need to take a different tack to simplify the
problem. Rather than allowing packets to live forever within
the network, we devise a mechanism to kill off aged packets
that are still hobbling about.
 Packet lifetime can be restricted to a known maximum
using one (or more) of the following techniques:
1. Restricted network design.
2. Putting a hop counter in each packet.
3. Time stamping each packet.
• TCP uses three-way handshake to establish connections in
the presence of delayed duplicate control segments as
shown in below figure.
Figure 4.5: Three protocol scenarios for establishing a connection using a three-way
handshake. CR denotes Connection Request. (a) Normal operation. (b) Old duplicate
connection request appearing out of nowhere. (c) Duplicate connection request and
duplicate ack.
3. Releasing a connection
Releasing a connection is easier than establishing one. There
are two styles of terminating a connection: asymmetric
release and symmetric release.
Asymmetric release is the way the telephone system works:
when one party hangs up, the connection is broken.
Symmetric release treats the connection as two separate
unidirectional connections and requires each one to be
released separately.
Asymmetric release is abrupt and may result in data loss.
Consider the scenario of below Fig. After the connection is
established, host 1 sends a segment that arrives properly at
host 2. Then host 1 sends another segment.
 Unfortunately, host 2 issues a DISCONNECT before the
second segment arrives. The result is that the connection is
released and data are lost.
Symmetric release does the job when each process has a
fixed amount of data to send and clearly knows when it has
sent it. In other situations, determining that all the work has
been done and the connection should be terminated is not so
obvious.
One can envision a protocol in which host 1 says ‘‘I am done.
Are you done too?’’ If host 2 responds: ‘‘I am done too.
Goodbye, the connection can be safely released.’’

Figure : Abrupt disconnection with loss of

data
Below Figure illustrates four scenarios of releasing using a
three-way handshake. While this protocol is not infallible, it
is usually adequate. In Fig. (a), we see the normal case in
which one of the users sends a DR (DISCONNECTION
REQUEST) segment to initiate the connection release.
When it arrives, the recipient sends back a DR segment and
starts a timer, just in case its DR is lost. When this DR
arrives, the original sender sends back an ACK segment and
releases the connection.
Finally, when the ACK segment arrives, the receiver also
releases the connection. Releasing a connection means that
the transport entity removes the information about the
connection from its table of currently open connections and
signals the connection’s owner (the transport user)
somehow.
Figure : Four protocol scenarios for releasing a connection. (a) Normal case of
three-way handshake. B) Final ACK lost. (c) Response lost. (d) Response lost and
subsequent DRs lost.
If the final ACK segment is lost, as shown in Fig. (b), the
situation is saved by the timer. When the timer expires, the
connection is released anyway. Now consider the case of the
second DR being lost.
The user initiating the disconnection will not receive the
expected response, will time out, and will start all over
again. In Fig. (c), we see how this works, assuming that the
second time no segments is lost and all segments are
delivered correctly and on time.
Our last scenario, Fig.(d), is the same as Fig.(c) except that
now we assume all the repeated attempts to retransmit the
DR also fail due to lost segments. After N retries, the sender
just gives up and releases the connection. Meanwhile, the
receiver times out and also exits.
4. Error control and Flow control:
– Error control is ensuring that the data is delivered with
the desired level of reliability, usually that all of the data
is delivered without any errors. Flow control is keeping a
fast transmitter from overrunning a slow receiver.
– In some ways the flow control problem in the transport
layer is the same as in the data link layer, but in other
ways it is different.
– The basic similarity is that in both layers a sliding
window or other scheme is needed on each connection to
keep a fast transmitter from overrunning a slow receiver.
– The main difference is that a router usually has relatively
few lines, whereas a host may have numerous
connections.
– This difference makes it impractical to implement the data
link buffering strategy in the transport layer.
– If the network service is unreliable, the sender must buffer
all TPDUs sent, just as in the data link layer.
– With reliable network service, other trade-offs become
possible.
– In particular, if the sender knows that the receiver
always has buffer space, it need not retain copies of the
TPDUs it sends.
– If the receiver cannot guarantee that every incoming
TPDU will be accepted, the sender will have to buffer
anyway.
– In the latter case, the sender cannot trust the network
layer's acknowledgement, because the
acknowledgement means only that the TPDU arrived,
Fig. 4.11: (a) Chained fixed-size buffers. (b) Chained
variable-sized buffers. C) One large circular buffer
per connection.
– Even if the receiver has agreed to do the buffering, there
still remains the question of the buffer size.
– If most TPDUs are nearly the same size, it is natural to
organize the buffers as a pool of identically- sized
buffers, with one TPDU per buffer, as in Fig.(a).
– If there is wide variation in TPDU size, a pool of fixed-
sized buffers presents problems.
– If the buffer size is chosen equal to the largest possible
TPDU, space will be wasted whenever a short TPDU
arrives.
– If the buffer size is chosen less than the maximum TPDU
size, multiple buffers will be needed for long TPDUs,
with the attendant complexity.
– Another approach to the buffer size problem is to use
variable-sized buffers, as in Fig. (b).
– The advantage here is better memory utilization, at the
price of more complicated buffer management.
– A third possibility is to dedicate a single large circular
buffer per connection, as in Fig. (c).
– This system also makes good use of memory, provided
that all connections are heavily loaded, but is poor if
some connections are lightly loaded.
5. Multiplexing:
– Multiplexing, or sharing several conversations over
connections, virtual circuits, and physical links plays a
role in several layers of the network architecture. In the
transport layer, the need for multiplexing can arise in a
number of ways. For example, if only one network
address is available on a host, all transport connections
on that machine have to use it.
– When a segment comes in, some way is needed to tell
which process to give it to. This situation, called
multiplexing, is shown in Fig. (a). In this figure, four
distinct transport connections all use the same network
connection (e.g., IP address) to the remote host.
Figure : (a) Multiplexing (b) Inverse Multiplexing
 Multiplexing can also be useful in the transport layer for
another reason. Suppose, for example, that a host has
multiple network paths that it can use. If a user needs more
bandwidth or more reliability than one of the network paths
can provide, a way out is to have a connection that
distributes the traffic among multiple network paths on a
round-robin basis, as indicated in Fig.(b).
6. Crash Recovery:
– If hosts and routers are subject to crashes or connections
are long-lived (e.g., large software or media downloads),
recovery from these crashes becomes an issue.
– If the transport entity is entirely within the hosts,
recovery from network and router crashes is
straightforward. The transport entities expect lost
segments all the time and know how to cope with them
by using retransmissions.
– A more trouble some problem is how to recover from
host crashes. In particular, it may be desirable for clients
to be able to continue working when servers crash and
quickly reboot.
 The Internet Transport Protocols(TCP, UDP)
 The Internet has two main protocols in the transport layer, a
connectionless protocol (UDP)and a connection-oriented
(TCP). The protocols complement each other.
I. UDP (User Datagram Protocol):
 it is a part of the TCP/IP protocol, so it is a standard protocol over the
internet. The UDP protocol allows the computer applications to send
the messages in the form of datagrams from one machine to another
machine over the Internet Protocol (IP) network.
 The UDP is an alternative communication protocol to the TCP
protocol (transmission control protocol). Like TCP.
 UDP provides a set of rules that governs how the data should be
exchanged over the internet. The UDP works by encapsulating the
data into the packet and providing its own header information to
the packet. Then, this UDP packet is encapsulated to the IP packet
and sent off to its destination.
 Both the TCP and UDP protocols send the data over the
internet protocol network, so it is also known as TCP/IP
and UDP/IP.
 There are many differences between these two
protocols. UDP enables the process to process
communication, whereas the TCP provides host to
host communication. Since UDP sends the messages
in the form of datagrams, it is considered the best-
effort mode of communication.
 TCP sends the individual packets, so it is a reliable
transport medium. Another difference is that the TCP is
a connection-oriented protocol whereas, the UDP is a
connectionless protocol as it does not require any
virtual circuit to transfer the data.
Features of UDP protocol:
 It is considered an unreliable protocol, and it is based on
best-effort delivery services. UDP provides no
acknowledgment mechanism, which means that the receiver
does not send the acknowledgment for the received packet,
and the sender also does not wait for the acknowledgment
for the packet that it has sent.
 Connectionless: The UDP is a connectionless protocol as it
does not create a virtual path to transfer the data. It does not
use the virtual path, so packets are sent in different paths
between the sender and the receiver, which leads to the loss
of packets or received out of order.
 Ordered delivery of data is not guaranteed: In the case of
UDP, the datagrams are sent in some order will be received
in the same order is not guaranteed as the datagrams are not
numbered.
 Ports: The UDP protocol uses different port numbers so
that the data can be sent to the correct destination. The port
numbers are defined between 0 and 1023.
 Faster transmission: UDP enables faster transmission as it
is a connectionless protocol, i.e., no virtual path is required
to transfer the data. But there is a chance that the individual
packet is lost, which affects the transmission quality. On the
other hand, if the packet is lost in TCP connection, that
packet will be resent, so it guarantees the delivery of the
data packets.
• Acknowledgment mechanism: The UDP does have any
acknowledgment mechanism, i.e., there is no handshaking
between the UDP sender and UDP receiver. If the message
is sent in TCP, then the receiver acknowledges that I am
ready, then the sender sends the data. In the case of TCP, the
handshaking occurs between the sender and the receiver,
whereas in UDP, there is no handshaking between the
sender and the receiver.
• Segments are handled independently: Each UDP segment
is handled individually of others as each segment takes
different path to reach the destination. The UDP segments
can be lost or delivered out of order to reach the destination
as there is no connection setup between the sender and the
receiver.
Why do we require the UDP protocol?
• As we know that the UDP is an unreliable protocol, but we
still require a UDP protocol in some cases. The UDP is
deployed where the packets require a large amount of
bandwidth along with the actual data. For example, in video
streaming, acknowledging thousands of packets is
troublesome and wastes a lot of bandwidth. In the case of
video streaming, the loss of some packets couldn't create a
problem, and it can also be ignored.
UDP Header Format
 In UDP, the header size is 8 bytes, and the packet size is
upto 65,535 bytes. But this packet size is not possible as the
data needs to be encapsulated in the IP datagram, and an IP
packet, the header size can be 20 bytes.
 therefore, the maximum of UDP would be 65,535 minus 20.
The size of the data that the UDP packet can carry would be
65,535 minus 28 as 8 bytes for the header of the UDP
packet and 20 bytes for IP header.
The UDP header contains four fields:
• Source port number: It is 16-bit information that identifies
which port is going t send the packet.
• Destination port number: It identifies which port is going to
accept the information. It is 16-bit information which is used to
identify application-level service on the destination machine.
• Length: It is 16-bit field that specifies the entire length of the
UDP packet that includes the header also. The minimum value
would be 8-byte as the size of the header is 8 bytes.
• Checksum: It is a 16-bits field, and it is an optional field. This
checksum field checks whether the information is accurate or
not as there is the possibility that the information can be
corrupted while transmission. In UDP, the checksum field is
applied to the entire packet, i.e., header as well as data part
whereas, in IP, the checksum field is applied to only the header
field.
Disadvantages:
 It provides an unreliable connection delivery service. It does
not provide any services of IP except that it provides
process-to-process communication.
 The UDP message can be lost, delayed, duplicated, or can
be out of order.
 It does not provide a reliable transport delivery service. It
does not provide any acknowledgment or flow control
mechanism. However, it does provide error control to some
extent.
Advantages
 It produces a minimal number of overheads.
Applications of UDP:
 Used for simple request-response communication when the
size of data is less and hence there is lesser concern about
flow and error control.
 It is a suitable protocol for multicasting as UDP supports
packet switching.
 UDP is used for some routing update protocols like
RIP(Routing Information Protocol).
 Normally used for real-time applications which can not
tolerate uneven delays between sections of a received
message.
 TCP (Transmission Control Protocol)
 UDP is a simple protocol and it has some very important
uses, such as client server interactions and multimedia, but
for most Internet applications, reliable, sequenced delivery
is needed. UDP cannot provide this, so another protocol is
required. It is called TCP.
 It is a transport layer protocol that facilitates the
transmission of packets from source to destination. It is a
connection-oriented protocol that means it establishes
the connection prior to the communication that occurs
between the computing devices in a network.
 This protocol is used with an IP protocol, so together,
they are referred to as a TCP/IP.
• The main functionality of the TCP is to take the data
from the application layer. Then it divides the data
into a several packets, provides numbering to these
packets, and finally transmits these packets to the
destination. The TCP, on the other side, will
reassemble the packets and transmits them to the
application layer.
• The transport layer has a critical role in providing
end-to-end communication to the directly
application processes. It creates 65,000 ports so that
the multiple applications can be accessed at the
same time. It takes the data from the upper layer,
and it divides the data into smaller packets and then
Features of TCP protocol
The following are the features of a TCP protocol:
• Reliable: TCP is a reliable protocol as it follows the flow
and error control mechanism. It also supports the
acknowledgment mechanism, which checks the state and
sound arrival of the data. In the acknowledgment
mechanism, the receiver sends either positive or negative
acknowledgment to the sender so that the sender can get to
know whether the data packet has been received or needs to
resend.
• Order of the data is maintained: This protocol ensures
that the data reaches the intended receiver in the same order
in which it is sent. It orders and numbers each segment so
that the TCP layer on the destination side can reassemble
them based on their ordering.
• Connection-oriented: It is a connection-oriented service
that means the data exchange occurs only after the
connection establishment. When the data transfer is
completed, then the connection will get terminated.
• Stream-oriented :TCP is a stream-oriented protocol as it
allows the sender to send the data in the form of a stream of
bytes and also allows the receiver to accept the data in the
form of a stream of bytes. TCP creates an environment in
which both the sender and receiver are connected by a
virtual circuit. This virtual circuit carries the stream of bytes
Working of TCP
 In TCP, the connection is established by using three-way
handshaking. The client sends the segment with its
sequence number. The server, in return, sends its segment
with its own sequence number as well as the
acknowledgement sequence, which is one more than the
client sequence number. When the client receives the
acknowledgment of its segment, then it sends the
acknowledgment to the server. In this way, the connection is
established between the client and the server.
Advantages of TCP
• It provides a connection-oriented reliable service, which
means that it guarantees the delivery of data packets. If the
data packet is lost across the network, then the TCP will
resend the lost packets.
• It provides a flow control mechanism using a sliding
window protocol.
• It provides error detection by using checksum and error
control by using Go Back or ARP protocol.
• It eliminates the congestion by using a network congestion
avoidance algorithm that includes various schemes such
as additive increase/multiplicative decrease (AIMD),
slow start, and congestion window.
Disadvantage of TCP
• It increases a large amount of overhead as each
segment gets its own TCP header, so fragmentation
by the router increases the overhead.
• TCP is made for Wide Area Networks, thus its size
can become an issue for small networks with low
resources.
• TCP runs several layers so it can slow down the
speed of the network.
TCP Header Format
• Source port: It defines the port of the application, which is
sending the data. So, this field contains the source port
address, which is 16 bits.
• Destination port: It defines the port of the application on the
receiving side. So, this field contains the destination port
address, which is 16 bits.
• Sequence number: This field contains the sequence number
of data bytes in a particular session.
• Acknowledgment number: When the ACK flag is set, then
this contains the next sequence number of the data byte and
works as an acknowledgment for the previous data received.
For example, if the receiver receives the segment number 'x',
then it responds 'x+1' as an acknowledgment number.
• HLEN: It specifies the length of the header indicated by the
4-byte words in the header. The size of the header lies
between 20 and 60 bytes. Therefore, the value of this field
would lie between 5 and 15.
• Reserved: It is a 4-bit field reserved for future use, and by
default, all are set to zero.
• Window size: It is a 16-bit field. It contains the size of data that
the receiver can accept. This field is used for the flow control
between the sender and receiver and also determines the amount
of buffer allocated by the receiver for a segment. The value of
this field is determined by the receiver.
• Flags
• There are six control bits or flags:
• URG: It represents an urgent pointer. If it is set, then the data is
processed urgently.
• ACK: If the ACK is set to 0, then it means that the data packet does
not contain an acknowledgment.
• PSH: If this field is set, then it requests the receiving device to push
the data to the receiving application without buffering it.
• RST: If it is set, then it requests to restart a connection.
• SYN: It is used to establish a connection between the hosts.
• FIN: It is used to release a connection, and no further data
exchange will happen.
•
• Checksum: It is a 16-bit field. This field is optional in UDP, but in
the case of TCP/IP, this field is mandatory.
• Urgent pointer: It is a pointer that points to the urgent data
byte if the URG flag is set to 1. It defines a value that will be
added to the sequence number to get the sequence number of
the last urgent byte.
• Options: It provides additional options. The optional field is
represented in 32-bits. If this field contains the data less than
32-bit, then padding is required to obtain the remaining bits.

These bits enable flow control, connection establishment and
termination, connection abortion, and the mode of data transfer in
TCP.

A TCP Connection
• TCP is connection-oriented. A connection-oriented transport protocol
• establishes a virtual path between the source and destination. All the
segments belonging to a message are then sent over this virtual path.
Using a single virtual pathway for the entire message facilitates the
acknowledgment process as well as retransmission of damaged or lost
frames.
• In TCP, connection-oriented transmission requires 2 phases:

1. connection establishment,

2. connection termination.
1. TCP connection establishment(3 way handshaking):
Step 1: SYN
 SYN is a segment sent by the client to the server. It acts as
a connection request between the client and server. It
informs the server that the client wants to establish a
connection.
 The client sends the first segment, a SYN segment, in
which only the SYN flag is set. NOTE:A SYN segment
cannot carry data, but it consumes one sequence number.
Step 2: SYN-ACK
 It is an SYN-ACK segment or an SYN + ACK segment
sent by the server. The ACK segment informs the client that
the server has received the connection request and it is
ready to build the connection. The SYN segment informs
NOTE: A SYN+ACK segment cannot carry data, but
does consume one sequence number.
Step 3: ACK
 ACK (Acknowledgment) is the last step before establishing
a successful TCP connection between the client and server.
The ACK segment is sent by the client as the response of
the received ACK and SN from the server. It results in the
establishment of a reliable data connection.
 After these three steps, the client and server are ready for
the data communication process. TCP connection and
termination are full-duplex, which means that the data
can travel in both the directions simultaneously.
 In the event that two hosts simultaneously attempt to
establish a connection between the same two sockets, the
sequence of events is as illustrated in Fig.(b). The result of
these events is that just one connection is established, not
two, because connections are identified by their end points.
If the first setup results in a connection identified by (x, y)
and the second one does too, only one table entry is made,
namely, for (x, y).
2. connection termination (A 4-way handshake)
Any device establishes a connection before
proceeding with the termination. TCP requires 3-
way handshake to establish a connection between
the client and server before sending the data.
Similarly, to terminate or stop the data transmission,
it requires a 4-way handshake. The segments
required for TCP termination are similar to the
segments to build a TCP connection (ACK and
SYN) except the FIN segment. The FIN segment
specifies a termination request sent by one device to
the other.
 The client is the data transmitter and the server is a receiver
in a data transmission process between the sender and
receiver. Consider the below TCP termination diagram that
shows the exchange of segments between the client and
server.
 The diagram of a successful TCP termination showing the
four handshakes is shown below:
Step 1: FIN
 FIN refers to the termination request sent by the client to
the server. The first FIN termination request is sent by the
client to the server. It depicts the start of the termination
process between the client and server.
Step 2: FIN_ACK_WAIT
 The client waits for the ACK of the FIN termination request
from the server. It is a waiting state for the client.
Step 3: ACK
 The server sends the ACK (Acknowledgement) segment
when it receives the FIN termination request. It depicts that
the server is ready to close and terminate the connection.
Step 4: FIN _WAIT_2
 The client waits for the FIN segment from the server. It is a
type of approved signal sent by the server that shows that
the server is ready to terminate the connection.
Step 5: FIN
 The FIN segment is now sent by the server to the client. It
is a confirmation signal that the server sends to the client. It
depicts the successful approval for the termination.
Step 6: ACK
 The client now sends the ACK (Acknowledgement)
segment to the server that it has received the FIN signal,
which is a signal from the server to terminate the
connection. As soon as the server receives the ACK
segment, it terminates the connection.
Flow Control or TCP Sliding Window
TCP Congestion Control
Tcp timer management